<div dir="ltr">Aurelien,<br><br>Thank you for your help. I enabled GST_DEBUG on gstrtpjitterbuffer and udpsrc. From the log, I can see part of the UDP packets are missing (there exists gap of the packet number between consecutive packets that are pushed into jitterbuffer). If I force rtspsrc to use TCP for all transmissions, then everything is perfect and no packet is dropped thanks to TCP's flow control and retransmission. Since I'm using Darwin Streaming Server, I guess the server is pretty aggresive in sending out UDP packet, my client board cannot catch up the incoming UDP packets and packet drop happens. Maybe I have to find out someway to control the sending rate at the server side when using UDP.<br>
<br>Best regards,<br>Felix<br><br><div class="gmail_quote">On Wed, Aug 13, 2008 at 2:33 AM, Aurelien Grimaud <span dir="ltr"><<a href="mailto:gstelzz@yahoo.fr" target="_blank">gstelzz@yahoo.fr</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><div>
Hi, there is already a gstrtpjitterbuffer inside rtspsrc, so no need for a second one.<br>
<br>You can run your gst-launch with GST_DEBUG=3 or 4 or 5 to track the received buffers.<br>Are you sure RTP is sent correctly ? received correctly ? <br>packets are lost in your network ? <br>network card fails ...<br>
<br>
Check sent packet with wireshark on sending host <a href="http://10.38.164.205" target="_blank">10.38.164.205</a>.<br>Check received packet with wireshark on receiving host.<br>wireshark will decode RTP, you will be able to see wether there are packet drops in your network or not.<br>
</div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><br>Initial file (water_mpeg4.mp4) contains only video ?<br>How do you find out that a quarter of the stream is received ?<br>Could remaining 3/4 be audio
?<br><br>You can use "rtpmp4vdepay ! mpeg4videoparse ! ffmux_mov ! filesink" to get a viewable file and see if those dropped buffers are really dropped.<br><br>Aurelien<br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;">
----- Message d'origine ----<br>De : Ting Wang <<a href="mailto:wangting@gmail.com" target="_blank">wangting@gmail.com</a>><br>À : <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a><br>
Envoyé le : Mardi, 12 Août 2008, 11h17mn 54s<br>Objet : [gst-devel] How can I get detailed information about those packets dropped by rtspsrc ?<div><div></div><div><br><br><div dir="ltr">Hi, <br><br>I'm using Gstreamer to setup a streaming video client using rtspsrc. Thanks to the help from Aurelien, I successfully setup the pipeline and it's rolling good. Now I find that with the packet drop is quite frequent even in locale area network. I have been tried to configure the parameter "latency" on rtspsrc/gstrtpjitterbuffer, but they didn't help. The video source rate is 788kbps, lasts 6 seconds.
By dumping the received file to disk, it turned out that only a quarter of the stream was successfully received ("gst-launch -v rtspsrc location=rtsp://<a rel="nofollow" href="http://10.38.164.205/water_mpeg4.mp4" target="_blank">10.38.164.205/water_mpeg4.mp4</a> ! gstrtpjitterbuffer latency=20000 ! rtpmp4vdepay ! filesink location=./stream.m4v"). Most of them was dropped for some reason I don't know. How can I get information regarding those drooped packets? Thank you for your advice :)<br>
<br>Best regards<br><br>-felix<br></div></div></div></div></div></div><div><br>
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